Hi there, Outgoing call centre: Sip-phones, => pbx1 =IAX2(trunk(s)=>vrouter=>Sip/Voip/provider
I'm experiencing asterisk crashes (1.4.3 on FreeBSD 6.2 + G729 codecs elsewhere downloaded), but the strange part is that I see this on the vrouter: 192.168.123.142 0720653046 3d9ef7915f5 00102/558531892 g729 No Rx: ACK 192.168.123.142 0720653046 4fc72983748 00102/558564206 g729 No Rx: ACK 192.168.123.142 0725285584 0daf511d483 00102/00000 unkn No Init: INVITE 192.168.123.142 0725285584 2f71adff548 00102/00000 unkn No Init: INVITE 192.168.123.142 0725285584 350f2426641 00102/00000 unkn No Init: INVITE 192.168.123.142 0725285584 6a5e17da29c 00102/00000 unkn No Init: INVITE 192.168.123.142 0725285584 6b85b3847f6 00102/00000 unkn No Init: INVITE Thus according to the "sip show channels" I have ~43 calls, but according to "iax2 show channels" I only see 14 calls. Is there something I'm doing wrong in the dialplan? (I just set the call recording on and Dial(Sip/Provider)) Or is this "expected"?? -- Hendrik Visage _______________________________________________ Asterisk-BSD mailing list Asterisk-BSD@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-bsd
